KELOMPOK PROYEK AKHIR SINYAL DAN SISTEM

1. Muh Ilham Akbar Iffagano

PROJECT 1

2. Reyfista Pangestu - Ahmad Salaam Mirfananda

PROJECT 3

3. Ubay M Noor - Amirsyah Rayhan M

PROJECT 6

4. Dita Tessa Parastika - Ismi Rosyiana Fitri

PROJECT 30

5. M. Audy F. - Muhammad Raditya Gumelar

PROJECT 25

6. Widi Destrianda - Benni Mustafa

PROJECT 27

7. Ginas Alvianingsih - Dwinanri Egyna

PROJECT 7

8. Muhammad Erfinza - Angga Hilman Hizrian

PROJECT 24

9. Andreas P. Aji - Ferdy Kurniawan

PROJECT 20

10. M.Hilmy Iskandar - Martino Adisuwono

PROJECT 19

11. Aiman Setiawan- Suharsono Halim

PROJECT 22

12. Muhammad Haekal - Daniel Moses

PROJECT 17

13. Ilham Dwi P - Harianto Adriprasetyo

PROJECT 29

14. Claudia Khansa - Faya Safirra

PROJECT 11

15. Dawud Shibghotulloh - Diamod Ravi

PROJECT 36

16. Jendra Riyan Dwiputra - Andik Suprayogi

PROJECT 9

17. Qashtalani Haramaini - Ilyasa Rafif

PROJECT 23

18. Fajar Tri Wardana - Rahmat Sigalingging

PROJECT 8

19. Agnes Grace S N - Susan Wiguna

PROJECT 5

20. Luthfan Fauzan - Restu Nugroho

PROJECT 33

21. Ahmad Dzul Faiq - M. Al Fatih

PROJECT 34

22. Samuel Zakaria - Ester Nugraheny N.P

PROJECT 18

23. Rachmat Romario Akbara - Yohan Binsar H.G

PROJECT 26

24. Rialdo Stefan Josua - Hirzi Hasan

PROJECT 12

25. Adhelia Irawan - Maulidya Falah

PROJECT 14

26. Chaizar Ali Fachrudien - Achyar Maulana Pratama

PROJECT 2

27. Erdi Nindito Rumono Samsudiat

PROJECT 28

28. Zaky Ramadhan - Antonius Listyo

PROJECT 10

29. Findal Darmaja - Josan Putra

PROJECT 16

30. Jodi Malikan - Bagus Setiawan

PROJECT 21

31. Ganang Rizky - Pradana Damara

PROJECT 37

32. Kenny Prasetyo Rizky Muhammad Reza

PROJECT 13

33. Andhika Kumara D - Ahmad Fauzi Arief

PROJECT 4

34. Budianto - Andre Jatmiko

PROJECT 32

35. Dara Azka Abi Iqbal Prasetyo

PROJECT 35

36. Erasmus NK Dhani Teja K.

PROJECT 15

37. Muhammad Iqbal Barry Muhammad

PROJECT 31

Pembagian Proyek dilaksanakan secara undi, bila perlu penjelasan hubungi

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SIGNALS AND SYSTEMS PROJECT 2014

General instructions:
1. Make a user-friendly GUI application program by using MATLAB
2. Generalize the input parameters in each application you make, if you are
given a specific value, you must be able to input some other values as well.

PROJECT 1
Generate analog and discrete signals and do some basic operations on those signals,
such as adding, subtracting, multiplying, flipping, delaying, etc.

PROJECT 2
Make an application that can be used to determine if a given system is LTI or not. In
particular, if the system is linear or not, if the system is time invariant or not.

PROJECT 3
Make an application that is bale to calculate the average power of continuous and
discrete time periodic signals. The input signals can be generated by using the wellknown functions or the arbitrary functions.

PROJECT 4
Write a MATLAB program to decompose a continuous or a discrete time signal into its
odd and even parts.

PROJECT 5
A Sample and Hold circuit is used to help the conversion of an analog signal to be
digital one. The circuit conducts the sample operation and followed by uniform
quantizer. You are required to simulate the digitalization process using MATLAB. Also,
demonstrate the effect of sampling frequency.

PROJECT 6
You are supposed to write a MATLAB program to plot the magnitude spectrum of
x t 3e0.2t cos10t for t 10 . Because this is an infinite length signal, you should
truncate the signal by choosing the proper length. Try to simulate and analyze the
effect of signal length and also the number of frequency sampling points. Also,
demonstrate the effect of windowing. In particular, you are asked to compare for
different window functions.

PROJECT 7
You are asked to simulate the impact of DFT length on resolution. For example,
consider a test signal x t sin 2 1000 t sin 2 1100 t sin 2 3000 t . You
generate an 0.0024 second sample of that signal using a sample rate if 10000 kHz.
You use rectangular or Hamming window to compute the windowed DFTs with three
lengths: 25, 128, and 1024. Your program is expected to answer these questions:
1. How do the rectangular window results compare to the hamming window results?
2. Are you able to see from the frequency response magnitude that the signal contains
three separate sinusoids? If so, for which DFT length can you distinguish the
sinusoids?
3. What does increasing the DFT length accomplish?

PROJECT 8
The same as PROJECT 7, but in this project you must evaluate the impact of window
length of resolution. In specific, your program is expected to answer these questions:
1. How do the rectangular window results compare to the Hamming window results?
2. Are you able to see form the frequency response magnitude that the signal contains
three separate sinusoids? If so, for which DFT length can you distinguish the
sinusoids?
3. What does increasing the length of the data window accomplish?

PROJECT 9
In this project, you asked to analyse the sidelobe levels and leakage of a window.
Consider two sinusoidal signals: x1 t sin 2 50 t and x2 t 0.031sin 2 77.5 t .
You sample the signals at a rate of 1000 Hz over an interval of 0.127 seconds. Let
x n x1 n x2 n . Use a 1024-DFT to compute the windowed transforms of the total
signal x n and x1 n . Compare the results for 128-point rectangular and 128-point
Hamming windows. Your program is expected to answer these questions:
1. Describe what you observe in these plots.
2. How do the sidelobe levels of the windows affect the results?
3. Which window(s) allow you to detect the presence of both sinusoids? Why?

PROJECT 10
An application of signals and systems course is an echo cancellation. In this project
youa re asked to demonstrate an echo cancellation system. Echos can result from
reflections of walls in a room or mismatched switching equipment in telephone lines.
The effect of a single echo can be modelled as y n x n x n N , where x n is
the voice signal, N is the delay, and is a constant. An echo removal system must
be described by difference equation of the echo equation above with input x n

replaced by y n , and output y n replaced by x n , which is equivalent to

y n x n y n N . In this simulation, try N 1000 and 0.5 .

PROJECT 11
In this project you will filter an audio signal. First, you should have an audio signal.
You design a 40th-order lowpass filter with cutoff frequency 0.125 f s (you can use fir1
command). Then, you pass both the left and right channel signals of your audio snippet
through the filter.

PROJECT 12
Using an audio signal, demonstrate the Nyquist sampling theorem.

PROJECT 13
In this project, you are asked to do a digital music synthesis: Beethovens Fifth
Symphony using MATLAB. Please consult
http://www.cse.yorku.ca/~mohammad/3451F13/docs/Lab3.pdf
Note that you can also compose other music!!!

PROJECT 14
Write a MATLAB program to show the correlation between two signals. In particular,
your program should be used to determine the periodicity of a signal by using the
autocorrelation property. Autocorrelation also has an application to quantify the effect
of noise on a periodic signal. Show this in your program.

PROJECT 15
You are asked to develop an understanding of how AM radio signals are modulated
and

demodulated.

Please

http://www.ece.tamu.edu/~hpfister/courses/ecen314/project2.pdf

consult
for

further

information.

PROJECT 16
Provide a program to analyse the frequency response of a causal discrete-time LTI
system implemented using the difference equation. For example, we have

y n 0.1x n 0.1176 x n 1 0.1x n 2 1.7119 y n 1 0.81y n 2

You are asked to plot H f . Also, provide an output signal if given an input signal,
for example x n cos 0.1 n u n .

PROJECT 17
Demonstrate the time shifting and frequency shifting properties of the DTFT.

PROJECT 18
You are asked to design a noise reduction system. Suppose you record a noisy sound
(signal) and then you sample it to be x n . You are asked to identify the spectrum
corresponds to the noise and speech signals by using DTFT. You can remove the
noise afterwards.

PROJECT 19
This project is similar to PROJECT 10, however, in this project you must estimate the
echo parameters. In a real problem you do not know the parameters: delay and
constant. Suppose you are given the data y n (which is corrupted by an echo) but
suppose you do not know the value of the delay N . Let ryy n be the autocorrelation
of

x n ,

y n , ryy n y n * y n and let rxx n be the autocorrelation of

rxx n x n * x n . You must first find a formula for the signal ryy n in terms of rxx n
.
PROJECT 20
Make a program to find the Fourier series coefficients for a periodic signal. For
example, x t cos t . Also, you should show the convergence of the Fourier series
of the periodic pulse signal. Sometimes the integration formula for the Fourier series
coefficient is difficult to carry out. In these cases, the coefficients can be obtained
numerically.

Remember

formula

f t dt T f mT .

For

example,

c k

1
T
x t e jk0t dt can be approximated as c k

T T
T

M 1

x mT e

jk0 m T

. You

m0

compare the results.

PROJECT 21
In this project, you will examine the effects of a low pass filter on speech. Write a
MATLAB script to read an audio file into Matlab. (wavread), filter the sound file with a
40th order low pass FIR filter with a cutoff of 1000 Hz. (fir1, filter), a note on using fir1
to generate the coefficients for the low-pass filter: you specify a cutoff frequency thats
been normalized to the Nyquist frequency = fs/2. Generate and plot the amplitude of
the transfer function of the filter in dB vs. frequency in Hz. (freqz) Play the audio file
after it has been sent through the filter. In addition to plotting the frequency response
of the filter, do the following plots: use fft to plot the frequency response of the original
signal, use fft to plot the frequency response of the filtered signal (should be very
enlightening), do amplitude vs. index plots of the signal, before and after filtering

PROJECT 22
You use high pass filter in PROJECT 21. Note that filter the sound file with a 40th
order high pass FIR filter with a cutoff of 3000 Hz. (fir1, filter)

PROJECT 23
Use a band pass filter to reduce noise in a set of simulated audio tones. Write a
MATLAB script to generate a sequence in MATLAB which samples the signal
f t

1
sin 2 f1t sin 2 f 2t sin 2 f3t , where
10

f1 440 ,

f 2 554.365 , and

f3 659.255 where it simulates the A-major chord. Generate a noisy musical chord

consisting of the sampled signal plus a noisy time-domain signal, you can use randn
in MATLAB. Filter the sound file with a 60th order band pass FIR filter with a cutoff of
f LO 0.9 f1 and f HI 1.1 f3 . Use fir1. Listen to the music before and after being filtered.

PROJECT 24
In this project you are asked to learn signal sampling, manipulation, and playback.
Please see
http://www.cse.yorku.ca/~mohammad/3451F13/docs/Lab2.pdf

PROJECT 25
In this project you should do Fourier sound synthesis. Please see
http://www.cse.yorku.ca/~mohammad/3451F13/docs/Lab4.pdf

PROJECT 26
You will create a signal with added high frequency noise
1. Typeload mtlb
2. You can hear a voice say "MATLAB." This is the signal to which you will add
noisesoundsc(mtlb,Fs)
3. Create a noise signal noise = cos(2*pi*3*Fs/8*(0:length(mtlb)-1)/Fs)';(You can
hear the noise signal by typing soundsc(noise,Fs))(You can also use random
function to introduce noise)
4. Add the noise to the original signalu = mtlb + noise;
5. Scale the signal with noiseu = u/max(abs(u));(You scale the signal to try to
avoid overflows later on. You can hear the scaled signal with noise by typing
soundsc(u,Fs))
6. Display the frequency spectrum using FFT
7. View the scaled signal with noisespecgram(u,256,Fs);colorbar(In the
spectrogram, you can see the noise signal as a horizontal line at about 2800
Hz,which is equal to 3*Fs/8).
8. Use low-pass filters to eliminate high frequencyb = ones(1,10)/10; % 10 point
averaging filterfy = filtfilt(b,1,x); % Noncausal filteringfyy = filter(b,1,x); %
Normal filtering
9. Use FFT to plot the power spectrum of the filter signal and compare it to both,
the originaland the corrupted signals.
You will be provided an ECG ( Electrocardiogram) signal with noise.Your task is to
create a matlab program to determine the frequency of the noise and eliminate
thenoise signal.

PROJECT 27
1. The periodic signal x(t) is shown below. Determine the fundamental period T 0.
Write a MATLAB code to plot x(t), using enough points to get "smooth" curve.
This is just reproducing the curve provided below.
2. Compute the Fourier series coefficients for x(t) (if you can find it in the book,
that is ok). Plot the single-sided and double-sided spectra up to the 10th
harmonic.
3. Plot partial sums of the Fourier series for x(t) (terms 1 through N in the infinite
series). In your opinion, what is the value of N that results in a "good"
reproduction of x(t).
Document your work, include plots to illustrate your work and your conclusions.
Include your MATLAB code.

PROJECT 28
Consider the following two signals:

1, | t | 3
x1 (t )
0, elsewhere

1, | t | 1
x 2 (t )
0, elsewhere
a) Plot these two signals on the same figure. Use a time axis of [-5,5]. Add labels, etc
b) Derive (mathematically) the Fourier Transforms X1(f) and X2(f) of these two
signals
c) Write a program to compute the Fourier Transforms numerically, and compare
with b)
Note: you can use the Matlab command trapz for numerical integration
c) Plot the magnitude spectrum for the signals on the same figure.
d) Discuss your results and explain the relationship between the spectra X1(f) and
X2(f).

PROJECT 29
The impulse response h(t) for a particular LTI system is shown below. All parts
of this lab make use of this h (t).
h(t) = [3e-3t + 5 e-2t + e-t (4 cos(3t)+ 6 sin(3t)) + e-4t] . u(t)
1. Plot the impulse response for h(t) directly from the above equation by creating
a time vector.
2. Use the residue function to determine the transfer function H(s). Determine the
locations of the poles and zeros of H(s) with the roots function, and plot them
in the s-plane (x for poles, o for zeros).
3. Use the freqs function to plot the magnitude and phase of the transfer function.
Document your work, include plots to illustrate your work and your
conclusions. Include your MATLAB code.

PROJECT 30
Consider the system block diagram shown below:

(a) Use the series function to find the numerator and denominator polynomial
coefficient vectors for the cascade connection of G1(s) and G2(s). Use the printsys
function to find the overall transfer function
() =

()
()

.
Hint: in the Matlab help, terms like SYS, SYS1, etc., refer to the specification
of a system by a pair of vectors that give the coefficients for the numerator and
denominator polynomials of the transfer function. For example,
>> numG1= [1 1];
>> denG1=[1 2];
>> numG2=[1];
>> denG2=[500 0 0];
>> [num,den]=series(numG2,denG2,numG1,denG1);
>> printsys(num,den);
(b) Use impulse to plot the system impulse response g(t). Note: impulse assumes
that the system is causal (ROC of the transfer function is a right half-plane).
(c) Use step to plot the system step response.
(d) Plot the signal
() = cos(5) ()
for t [0, 5] using a time resolution of 0.002 sec. Use lsim with a left-hand
argument to plot the system response for t [0, 5] when r(t) is the system input.
Use a time resolution of 0.002 sec for your plot.
(e) Use pzmapwith left-hand arguments to observe the poles and zeros of G(s). Is
the system stable? (Justify your answer).
(f) Usepzmapwithout left-hand arguments to generate a pole-zero plot for G(s).

PROJECT 31
Consider the feedback system depicted in the block diagram below:

Note that the gain of the feedback path is unity (1).

(a) Use feedback to find the numerator and denominator polynomial coefficient
vectors of the closed-loop transfer function
() =

()
()

For feedback, you will need the numerator and denominator polynomial
coefficient vectors for the series connection of G1(s) and G2(s) that you
obtained in the above (problem (1)). The transfer function of the feedback
path in the system block diagram above is equal to one, so you can describe
it with the numerator polynomial coefficient vector [1] and the denominator
polynomial coefficient vector [1].
(b) Use printsysto find G(s).
(c) Use impulseto plot the system impulse response g(t). Give a brief
qualitativedescription of the main effect of adding feedback to this system as
compared tothe system in problem (1).
(d) Use pzmapwith left-hand arguments to observe the poles and zeros of G(s).
Isthe system stable? (Justify your answer).
(e) Use pzmapwithout left-hand arguments to generate a pole-zero plot for G(s).
(f) Plot the signal

() = 10000 cos (
) ()
500
for t [0, 20000] using a time resolution of 10 sec. Use lsim to plot the system
response when r(t) is the system input for t [0, 20000] using a time
resolution of 10 sec.

PROJECT 32
Consider the feedback system depicted in the block diagram below:

(a) Use feedback to find the numerator and denominator polynomial

coefficient vectors of the closed-loop transfer function
() =

()
()

(b) Use printsys to find G(s).

(c) Use impulse to plot the system impulse response g(t).
(d) Use pzmap with left-hand arguments to observe the poles and zeros of
G(s). Is itcausal? Is the system stable? (Justify your answers).
(e) Use pzmap without left-hand arguments to generate a pole-zero plot for
G(s).

PROJECT 33
H is a causal discrete-time LTI system with input x[n] and output y[n] related by the
linear constant coefficients difference equation
y[n] = 0.6y[n 1] + x[n]
(a) Find the transfer function H(z) analytically (using paper and pencil).
Use zplane to generate a pole-zero plot for H(z). Is the system stable?
(Justify your answer).
Note:You should get the vector [1] for the numerator polynomial
coefficientvector of H(z). On some versions of Matlab, this will create
an incorrect polezeroplot; the plot may show an incorrect zero at z = 1.
If this happens, you canfix it by using [1 0] for the numerator coefficient
vector. The vectors [1] and [1 0]actually specify the same numerator for
H(z), since 1 = 1 + 0z1.
(b) Use freqzto plot the magnitude |H(ej!)| and phase ]H(ej!) of the system
frequency response ( )using N = 1024 points.
(c) Use impzto plot the system impulse response h[n].

PROJECT 34
Second Order Response
Consider a transfer function of the following form corresponding to a right-sided
response.
() =

+ 2[] + ||2

The pole p is complex. Let =

For this question, r=0.9 and = /2.
(a) Use the Matlab function zplane to plot the poles and zeros of H(z) .
(b) Use freqz to calculate the complex frequency response. Plot the magnitude
and phase of thefrequency response corresponding to H(z) . Categorize this
response in terms of is it lowpass,highpass, bandpass, or bandstop.
(c) The group delay of a response is given by
()
() =

A good group delay is one which is constant with frequency. What does this
constancy imply for thephase? Why would filter designers prefer to deal with
group delay rather than phase response? Use the Matlab function grpdelay to
get the group delay associated with the filter given above. Plot the group
delay. What are the units of group delay? Label the axis appropriately.
(d) Plot the impulse response of the filter (first 20 or so samples).

PROJECT 35
DTFT for FIR signals
We have been using the routine freqz to compute frequency responses. In this
problem you will write your own DTFT routine for finite length sequences. The DTFT
will be of the form
function H = DTFTFIR (h, n, w)
nw = length(w);
nh = length(h);
for (k = 1:nw)
H(k) =
end
return
The input to this routine is a set of response values (h) with corresponding values of
time indices (n), and a vector of frequencies for which the response is to be
evaluated. Test your routine against freqz using the following two sets of responses
(use frequencies from 0 to in, say, 100 steps). Plot the magnitude response and
the phase response for each case.
(a)
h = [4 3 2 1 1 2 3 4];
n = -2:5;
Explain the shape of the phase response curve specifically, why are there
jumps in the function? Hint: where do the jumps occur with respect to the
magnitude curve?
(b)
n = 0:20;
h = n .* (0.9).^n;

PROJECT 36
DTFT for IIR signals
We have been using the routine freqz to compute frequency responses. In this
problem you will write your own DTFT routine for infinite length sequences specified
by a numerator polynomial and a denominator polynomial (polynomials in 1 z ). The
DTFT will be of the form
function H = DTFTIIR (b, a, w)
nw = length(w);
nb = length(b);
na = length(a);
for (k = 1:nw)
H(k) =
end
return
To do this first consider the rational polynomial
() =

() 0 + 1 1 + +
=
() 0 + 1 1 + +

It is then clear that the frequency response corresponding to H(z) for a series of
frequencies is the point-bypoint division of the frequency response of B(z) divided by
the frequency response of A(z) .
Write your DTFT routine and test it on the following responses. Plot the magnitude
and phase for each, comparing the results with those from freqz.
(a) [] = (0.5) ()
1 + 1 + 2 + 3

(b) () = 10.18 1 + 0.81 2

PROJECT 37
The following program can be used to verify the time shifting property of
discrete-time
Fourier transform(DTFT):
clf;
w = -pi:2*pi/255:pi; wo = 0.4*pi; D = 10;
num = [1 2 3 4 5 6 7 8 9];
h1 = freqz(num, 1, w);
h2 = freqz([zeros(1,D) num], 1, w);
subplot(2,2,1)
plot(w/pi,abs(h1));grid
title('Magnitude Spectrum of Original Sequence')
subplot(2,2,2)
plot(w/pi,abs(h2));grid
title('Magnitude Spectrum of Time-Shifted Sequence')
subplot(2,2,3)
plot(w/pi,angle(h1));grid
title('Phase Spectrum of Original Sequence')
subplot(2,2,4)
plot(w/pi,angle(h2));grid
title('Phase Spectrum of Time-Shifted Sequence')
Q.1 which parameter controls the amount of time-shift?
Q.2 Repeat the above program for signal in 5.21 and for -5<=n<=5, and time shift
value +5.
Q.3 describe the function of freqz.
Q.4 Write a program to show the frequency-shifting of DFT for following signal and
frequency-shift value of 0.4*pi (i.e., wo= 0.4*pi).